Our invention relates to digital speech communication and, more particularly, to arrangements for reducing the transmission rate for digital speech communication.
In speech communication systems, it is often desired to transmit a speech signal in digital form to provide secure communication or to improve the intelligibility of the signal in the presence of noise. Analog-to-digital conversion of the speech signal generally requires the signal to be sampled at a rate that is twice the highest frequency component of the analog signal. A voice signal may be reproduced in a band from 200 to 3000 Hz so that a sampling rate of 6000 Hz or greater is needed. In coding the sampled signal by means of one or more pulse code modulation techniques, a plurality of bits are produced responsive to the magnitude and sign of each sample. Consequently, the bit rate transmitted is substantially greater than the sampling rate. If the number of bits for each sample is reduced to limit the bit rate, the resulting signal is correspondingly degraded, and quantizing noise from the modulation process further increases the degradation of the signal reconstructed from the pulse code. The quantizing degradation spans and affects the entire frequency range of the original speech signal.
As disclosed in U.S. Pat. No. 3,674,939, issued July 4, 1972, to F. A. Brooks, and elsewhere, a reduction of the required bit rate of a digital speech communication system may be effected by dividing the speech spectrum uniformly into a plurality of subbands each of which is transposed by modulation techniques into a common, relatively narrow baseband. In this manner, the sampling rate required for the transmission of transposed baseband signals is reduced to correspond to the highest frequency of the common baseband. Separate coding of each baseband signal, however, multiplies the bit rate so that little or no reduction of overall bit rate is achieved without limiting the rate of the speech signal or otherwise affecting intelligibility. A replica of the analog speech signal is produced by retransposition of the separated decoded baseband components and summation of the retransposed subbands.
Sampling with aliasing for efficient frequency transposition whereby a frequency band W extending from frequency f.sub.1 to frequency f.sub.1 +W can be transposed to a baseband by digitally sampling it at a rate equal to 2W samples per second has been suggested by C. B. Feldman and W. R. Bennett in their article entitled "Bandwidth and Transmission Performance" appearing in the Bell System Technical Journal, Volume 28, No. 3, pages 594-595, July, 1949.
It is known, as disclosed in the article, "The Design of Speech Communication Systems," by Leo L. Beranek, appearing in the Proceedings of the IRE, September 1947, on pages 880-884, that the speech spectrum can be analyzed in terms of the ability of a communication system to transmit speech intelligibly in the presence of noise. On the basis of experimentally performed articulation tests, an Articulation Index has been developed which relates the contribution of frequency bands of the speech spectrum to speech signal intelligibility. The analysis shows that for the spectrum of speech, a frequency band of given width contributes differently to intelligibility depending upon its frequency location in the spectrum. Consequently, a uniform division of the speech spectrum into equal width subbands in accordance with the prior art for digital encoding and transmission of a speech signal may result in a relatively low intelligibility replica of the original speech signal. It is an object of the invention to provide an economical digital speech communication arrangement adapted to minimize the transmitted bit rate in accordance with predetermined intelligibility standards.